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gerrit-public.fairphone.software / fp2-dev / platform / external / chromium_org / third_party / webrtc / 3ded5808c2dec22ec97b3b0c855f1bcb75c1db09
  1. 3ded580 Update makefiles after merge of Chromium at 281279 by Android Chromium Automerger · 10 years ago
  2. c7343a3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at d13c3753199496aeddc73ec88548da73283c312f by Android Chromium Automerger · 10 years ago
  3. d13c375 Implement BUILD.gn for desktop_capture. by jiayl@webrtc.org · 10 years ago
  4. 4fa5c51 Make deadlock suppressions less generic. by andresp@webrtc.org · 10 years ago
  5. 65a971a Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. by andresp@webrtc.org · 10 years ago
  6. 2836899 Add tkchin@ to OWNERS. by tkchin@webrtc.org · 10 years ago
  7. 9113f0a webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 by henrike@webrtc.org · 10 years ago
  8. 34c5b23 Fix compile error introduced with r6571. by stefan@webrtc.org · 10 years ago
  9. 96583a9 Fixes a potential BWE clock mismatch bug. by stefan@webrtc.org · 10 years ago
  10. 2220b7a audio_processing/aec: Refactors NonLinearProcessing to prepare for NEON optimizations by bjornv@webrtc.org · 10 years ago
  11. 222d84a Use X509_NAME, not struct X509_name_st. by henrike@webrtc.org · 10 years ago
  12. 96a2a61 Neon version of cftmdl_128() by bjornv@webrtc.org · 10 years ago
  13. 8f02f89 Add ExperimentalNs support in Config by aluebs@webrtc.org · 10 years ago
  14. 88b558f Reserve RTP/RTCP modules in SetSSRC. by pbos@webrtc.org · 10 years ago
  15. b9cd54f Neon version of cft1st_128() by bjornv@webrtc.org · 10 years ago
  16. 8ade059 Removing W3C conformance tests after move to web-platform-tests. by phoglund@webrtc.org · 10 years ago
  17. 983db6a Make MediaOptimization thread-safe. by wuchengli@chromium.org · 10 years ago
  18. ec28212 GN: Fix build by disabling compiler warning in base. by kjellander@webrtc.org · 10 years ago
  19. b94f847 GN: Refactor base/BUILD.gn and fix dbus-glib error. by kjellander@webrtc.org · 10 years ago
  20. 484a4e7 Rebase webrtc/base with r6555 version of talk/base: by henrike@webrtc.org · 10 years ago
  21. d505b21 constructormagic.h macros are duplicated in several repositories. undef them in webrtc to prevent conflict for some build configurations. by henrike@webrtc.org · 10 years ago
  22. 900e8cf TSan: Move suppressions to source file. by kjellander@webrtc.org · 10 years ago
  23. eb67a6b Refactor Call-based tests. by pbos@webrtc.org · 10 years ago
  24. 958883a Receiver bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 10 years ago
  25. b3f0584 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 9ff0df06bd431ddbf595620f94ae515bbdcde2da by Android Chromium Automerger · 10 years ago
  26. 1e90eed clock.h: Removed GUARDED_BY annotation as it breaks som builds. by henrike@webrtc.org · 10 years ago
  27. a622b06 Don't forward declare RWLockWrapper in clock.h by henrik.lundin@webrtc.org · 10 years ago
  28. 9ff0df0 Fixes a bug causing NACKs to be dropped excessively at the send-side. by stefan@webrtc.org · 10 years ago
  29. 07737de Bump version number to 3.55 by tnakamura@webrtc.org · 10 years ago
  30. 9702d56 fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. by henrike@webrtc.org · 10 years ago
  31. 89740e1 pkg-config-wrapper should not be run when build_nss is disabled (=0). by henrike@webrtc.org · 10 years ago
  32. 841f8c8 Update makefiles after merge of Chromium at 279716 by Android Chromium Automerger · 10 years ago
  33. 4c21d3a Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c34b9e5d5cd44c31c4f9da649b71d0d3132cf516 by Android Chromium Automerger · 10 years ago
  34. 7eec1dd Add RTCP packet types to packet builder: by asapersson@webrtc.org · 10 years ago
  35. b5272df This is to compare NetEq with various codecs under a shared packet loss pattern. by minyue@webrtc.org · 10 years ago
  36. ccbe08e Neon version of FilterFar() by bjornv@webrtc.org · 10 years ago
  37. f1a6eac Remove payload duplication in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
  38. c34b9e5 Removing neteq decode lock and friends by henrik.lundin@webrtc.org · 10 years ago
  39. 26f68fe Neon version of ScaleErrorSignal() by bjornv@webrtc.org · 10 years ago
  40. b0cf5e1 Update makefiles after merge of Chromium at 278856 by Torne (Richard Coles) · 10 years ago
  41. 47b4b9d Annotating the rest of AcmGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  42. c782170 Fix array declarations in aec_core.c by andrew@webrtc.org · 10 years ago
  43. a2c6918 Annotating the rest of AudioCodingModuleImpl by henrik.lundin@webrtc.org · 10 years ago
  44. 3610f63 GN: Add BUILD.gn files + kjellander to OWNERS by kjellander@webrtc.org · 10 years ago
  45. ba4cfb1 Rebase webrtc/base with r6521 version of talk/base: by henrike@webrtc.org · 10 years ago
  46. e4b41b1 Disables tests that breaks Android bots by bjornv@webrtc.org · 10 years ago
  47. d474b29 Roll chromium_revision 272489:277350 + fix sanitizer options by kjellander@webrtc.org · 10 years ago
  48. ac5cd56 GN: BUILD.gn for system_wrappers by kjellander@webrtc.org · 10 years ago
  49. ab52e9a - Exit from a camera thread lopper loop() method only after all camera release calls are completed. This fixes camera exceptions observed from time to time when calling camera functions on a terminated looper. by glaznev@webrtc.org · 10 years ago
  50. 8c0544d Do not hold the critical section in VideoCaptureAndroid::SetCaptureRotation since it would case possible deadlock with OS Camear thread. by braveyao@webrtc.org · 10 years ago
  51. 4ee6348 Add tests of texture frames in video_send_stream_test. by wuchengli@chromium.org · 10 years ago
  52. e5c55da Do not call CaptureCursor in ScreenCapturerWinGdi if no MouseShapeObserver. by jiayl@webrtc.org · 10 years ago
  53. c497bcd Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 68f4c7b51ec6434b302de9e97ee01f5ccdb48aa2 by Android Chromium Automerger · 10 years ago
  54. 68f4c7b Revert 6481 and 6482 by fgalligan@google.com · 10 years ago
  55. c7a2c99 Maintain constantness of the input to iSAC-fix decoder, and prevent heap-buffer overflow. by turaj@webrtc.org · 10 years ago
  56. 035f764 Adding an empty constructor implementation to the AudioSink class by henrik.lundin@webrtc.org · 10 years ago
  57. 8cda294 Changes to tests and tools in audio_processing. by bjornv@webrtc.org · 10 years ago
  58. b9bd8c8 Ensure that the start bitrate can be set multiple times. by stefan@webrtc.org · 10 years ago
  59. 5c69a9f Adding test::AudioSink interface and derived classes by henrik.lundin@webrtc.org · 10 years ago
  60. 341671f Fixes and re-enables tests disabled on Android by bjornv@webrtc.org · 10 years ago
  61. ad3bcf4 Update makefiles after merge of Chromium at 278252 by Android Chromium Automerger · 10 years ago
  62. 852ce03 Update webrtc to fix unpack_lib expansion. by fgalligan@google.com · 10 years ago
  63. dfaba91 Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  64. 685eb96 Neon version of FilterAdaptation() by bjornv@webrtc.org · 10 years ago
  65. 82f4b96 Update PacketSource and RtpFileSource by henrik.lundin@webrtc.org · 10 years ago
  66. 743f486 Revert "Restore ptypes.txt file" by henrik.lundin@webrtc.org · 10 years ago
  67. 79ceb8c Revert 6473 "Update generated asm offsets scripts." by turaj@webrtc.org · 10 years ago
  68. 6f1646c Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  69. 7e87822 Add round-robin selection of send stream to pad on. by stefan@webrtc.org · 10 years ago
  70. 50d455e Add high perf mode to VP8 by niklas.enbom@webrtc.org · 10 years ago
  71. a487491 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ by henrike@webrtc.org · 10 years ago
  72. fd6b7a5 Rebase webrtc/base with r6464 version of talk/base: by henrike@webrtc.org · 10 years ago
  73. 28f69bb Revert 6458 "Since NetEq4 is ready to handle 48 kHz codec, it is..." by minyue@webrtc.org · 10 years ago
  74. d6d5bff Initial GN work for WebRTC by kjellander@webrtc.org · 10 years ago
  75. e78505f Restore ptypes.txt file by henrik.lundin@webrtc.org · 10 years ago
  76. 38a2d46 Updated W3C getusermedia tests to the latest version of the spec. by phoglund@webrtc.org · 10 years ago
  77. ab22857 Since NetEq4 is ready to handle 48 kHz codec, it is good to remove the 48-to-32kHz downsampling of Opus output. This facilitates webrtc to make full use of Opus's bandwidth and eliminates unneeded computation in resampling. by minyue@webrtc.org · 10 years ago
  78. f0cf127 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. by henrike@webrtc.org · 10 years ago
  79. 5c28a0a Update makefiles after merge of Chromium at 277521 by Android Chromium Automerger · 10 years ago
  80. e5a0f26 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 847dfa535730a30d57cf26d788d31070b70a02af by Android Chromium Automerger · 10 years ago
  81. cb4fdd1 Update makefiles after merge of Chromium at 277428 by Android Chromium Automerger · 10 years ago
  82. c7fcada Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 5fcef2b6df45ceab39ee96a616ab0a4d3c63b83a by Android Chromium Automerger · 10 years ago
  83. eddcc63 Add max limit of number for overuses. When limit is reached always apply the rampup delay. by asapersson@webrtc.org · 10 years ago
  84. 847dfa5 Add SDES, APP, IJ, SLI and PLI packet types to RTCP packet class. by asapersson@webrtc.org · 10 years ago
  85. e82b71d Remove ivinnichenko from webrtc/test/OWNERS by kjellander@webrtc.org · 10 years ago
  86. d3a2886 Importing ThreadChecker class from Chromium by henrik.lundin@webrtc.org · 10 years ago
  87. d998689 Adds aluebs@webrtc.org as owner to audio_processing by bjornv@webrtc.org · 10 years ago
  88. c1a2a43 common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16 by bjornv@webrtc.org · 10 years ago
  89. f89ce46 Implements start bitrate for new video API. by mflodman@webrtc.org · 10 years ago
  90. 0c14539 Add thread annotations to parts of ACMGenericCodec by henrik.lundin@webrtc.org · 10 years ago
  91. d05de74 Add missing sources to webrtc/base/base.gyp by kjellander@webrtc.org · 10 years ago
  92. bd98cef Add glaznev@ to OWNERS for webrtc/modules/video_capture and talk/app/webrtc. by glaznev@webrtc.org · 10 years ago
  93. 9257c64 Neon version of OverdriveAndSuppress() by bjornv@webrtc.org · 10 years ago
  94. 00dffd7 Pass GYP DEPTH variable to isolate. by kjellander@webrtc.org · 10 years ago
  95. dd32ef8 Revert 6415 "Update generated asm offsets scripts." by wu@webrtc.org · 10 years ago
  96. 555f957 json.h include different header files depending on WEBRTC_CHROMIUM_BUILD being defined or not. Since json.h/cc is not even used in chromium it is the wrong flag to use. Instead add WEBRTC_EXTERNAL define. Also added OWNERS for base which is a copy of system_wrappers owners as the two folders are being merged. by henrike@webrtc.org · 10 years ago
  97. 19f89a1 Enable pacing by default and remove the option to disable it from the new API. by stefan@webrtc.org · 10 years ago
  98. 0e43e6f Update generated asm offsets scripts. by fgalligan@google.com · 10 years ago
  99. 5fcef2b Revert 6411 "Revert 6407 "Revert 6405 "Update generated asm offs..." by kjellander@webrtc.org · 10 years ago
  100. 4150d6e Revert 6407 "Revert 6405 "Update generated asm offsets scripts."" by minyue@webrtc.org · 10 years ago